Re: [Asterisk-Users] Hangup Issues on TDM40B FXO Australia
I had a similar issue both with the X100P clones and TDM400. Both were fixed by enabling AU zone and the busydetect functions. Don't forget a full asterisk reload needs to take place after changing Zap conf files, not just a soft-reload. Best way is to reboot the computer. Mike I have a similar issue. I have 2 pstn lines and a phone plugged into my tdm400. If I make a call to the outside using the phone, and the pstn number is engaged, and I hang up, the line is not freed. I have been restarting asterisk to get my external line back. This does not happen if I make the same call from my pc (using sj phone). Malcolm [EMAIL PROTECTED] wrote: Afternoon all, After doing some test on my asterisk box I can successfully receive calls to my Asterisk PBX to a SIP phone from The Telstra PSTN network Dial out from a sip phone is also not an issue, all calls connect and terminate normally. If I call the Asterisk PBX say from PSTN in Zap1-1 and out through ZAP2-2 back to the PSTN (after entering the correct pin off course) the card does not appear to detect the hang-up, I then have to issues a soft hang-up to close the call, I presume this indicates the card is configured to receive the correct hangup signal I have tried enabling callprogress, busydetect and a few settings on the busycount but to no success I've also tried LS and KS signalling Does anyone else have any suggestions to get this to work with Australia's Telstra? Regards Haydn This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the originator of the message. This footer also confirms that this email message has been scanned for the presence of computer viruses. Any views expressed in this message are those of the individual sender, except where the sender specifies and with authority, states them to be the views of LMC. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.15 - Release Date: 22/05/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] oh323 on @homeasterisk
Can you please detail the steps you have taken to successfully compile this on @home asterisk? Regards Mike - Original Message - From: CM Rahman Jr. [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Saturday, April 09, 2005 4:09 PM Subject: [Asterisk-Users] oh323 on @homeasterisk Anybody here added oh323 to @homeasterisk? I have compiled and add the oh323. I am wondering if anybody able to add the oh323 under web interface AMP? If anybody did it or know how to do it, please let me know. It has option for sip, IAX.. why not add h323 !! Thanks ** C.M. Rahman Jr. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.5 - Release Date: 7/04/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Set system time over the phone
No LAN what-so-ever. Customer is very paranoid.
Yes, sanitisation would be handy. Perhaps I should call an AGI file to do
this. Although I'm not sure how you can hack a system using only numbers
0-9, # and *. I'm sure there's a way!!!
- Original Message -
From: Tzafrir Cohen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, April 05, 2005 8:02 PM
Subject: Re: [Asterisk-Users] Set system time over the phone
On Tue, Apr 05, 2005 at 09:45:54AM +1000, Mike Sander wrote:
I have installed Asterisk using the [EMAIL PROTECTED] image for a client that is
VoIP-a-phobic.
Hence the system cannot be connected to their LAN at all - don't ask why!
Does it have a lan connection at all? If so, you could use ntpd. Setting
the clock manually can have some side-effects and some services may
require running.
I have tested the clock at my installation lab, and all is fine, but they
might want to set/check it.
I know there is the SayUnixTime command, and it works fine to say the
time.
Is there a good dialplan command to test it? Best I've come across is
System, but this exits non-zero. Any ideas?
exten 456,1,Background(Please-set-time-mmddhhmm)
exten _.,1,System (date ${EXTEN})
Before passing input blindly to system, you need to sanitize it. E.g:
Any chance someone could dial a ';'? If so, that one can run an
arbitrary shell command (as Asterisk's user).
--
Tzafrir Cohen | New signature for new address and | VIM is
http://tzafrir.org.il | new homepage | a Mutt's
[EMAIL PROTECTED] || best
ICQ# 16849755 | Space reserved for other protocols | friend
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Re: [Asterisk-Users] Set system time over the phone
Looks good - thanks for the help!
Mike
- Original Message -
From: Roman Volf [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, April 05, 2005 4:48 PM
Subject: Re: [Asterisk-Users] Set system time over the phone
Another way is to do:
exten 456,1,Background(Please-set-time-mmddhhmm)
exten _.,1,System (echo ${EXTEN} /tmp/datetime )
Then have a cron job that runs every minute to check if file exists. For
example:
#!/bin/bash
if [ -f /tmp/datetime ] then
date `cat /tmp/datetime`
rm -f /tmp/datetime
fi
This should work fine.
Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]
Matt Riddell wrote:
Peter Bowyer wrote:
exten 456,1,Background(Please-set-time-mmddhhmm)
exten _.,1,System (date ${EXTEN})
If I dial 456 I get the message, so I type 04021305 (2nd April, 13:05).
On the console Asterisk reports the command Dial 04021305 exits
non-zero.
You need 'Read' instead of 'Background'.
No, because his next line is _.,1 so it will actually use the extension.
His problem is just one of permissions. Maybe he should use a suid prog
to set the date.
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[Asterisk-Users] Set system time over the phone
I have installed Asterisk using the [EMAIL PROTECTED] image for a client that is
VoIP-a-phobic.
Hence the system cannot be connected to their LAN at all - don't ask why!
I have tested the clock at my installation lab, and all is fine, but they
might want to set/check it.
I know there is the SayUnixTime command, and it works fine to say the time.
Is there a good dialplan command to test it? Best I've come across is
System, but this exits non-zero. Any ideas?
exten 456,1,Background(Please-set-time-mmddhhmm)
exten _.,1,System (date ${EXTEN})
If I dial 456 I get the message, so I type 04021305 (2nd April, 13:05).
On the console Asterisk reports the command Dial 04021305 exits non-zero.
If I then copy/paste into the shell, the command works.
Is there some weird brackets or something the System command is expecting
- the voip-info.org is up and down a lot at the mo.
Thanks
Mike
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[Asterisk-Users] Asterisk@Home H323
I am looking for a step-by-step on adding H323 to [EMAIL PROTECTED] So far I have installed [EMAIL PROTECTED], upgraded to the CVS-HEAD and followed instructions according to voip-info and this list's archives. I keep getting critical errors on compilation of H323, both Open 323 and OH323. Has anyone managed to install H323 with [EMAIL PROTECTED] If so, what steps did you perform. With Thanks Mike - Original Message - From: Matt [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 29, 2005 11:41 AM Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] 0.7 released Web Meetme is now installed by default and the meetme2 application is no longer needed. What does this mean exactly? Does this use the regular meetme as opposed to the meetme2 we had to setup before? On Mon, 28 Mar 2005 17:35:37 -0800 (PST), [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: We had added a lot to this release to our one button install of Asterisk. Now you can have even more features automatically installed and configured. Asterisk 1.0.7 AMP 1-10-007 Flash Operator Panel 0.20 Redesigned WebMeetme weather agi scripts Midnight Commander We have added some of our most requested features. - Web Meetme is now installed by default and the meetme2 application is no longer needed. - we now have ZAP extension thanks to AMP 007 - weather.agi reads the current weather report using text to speech __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to do something random?
Logically, you should build something like this:
1. Pick a number between 1 and 3
2. Save the number to a variable indicating which line you are about to try
3. Check if it's free, if so make a call
4. If not, pick a number between 1 and 2
5. Make sure you haven't tried this number before (a loop and perhaps an
array of line numbers)
6. When you find a not-yet-tried number, check if it's free. If so, make a
call
7. If not, loop again to find the remaining number, check if free, if so
make a call.
8. If you get here, all lines are busy - play the busy tone.
I'm sorry my coding is not up to scratch, but this seems like a good
application for an AGI script as it can do arrays and looping easier, and
you could build this up to many lines.
Mike
Take a look at the Random() command.
MARK.
Ronald Wiplinger wrote:
I want to change the below lines:
exten = _011.,1,SetGroup(line1); set current group to
line
exten = _011.,2,CheckGroup(1); check line1 does
not have more than 1
exten = _011.,3,Dial,SIP/[EMAIL PROTECTED]; use line-1
exten = _011.,103,1,SetGroup(line2); set current
group to line
exten = _011.,104,CheckGroup(1); check line2 does not
have more than 1
exten = _011.,105,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}; use line-2
exten = _011.,205,1,SetGroup(line3); set current
group to line
exten = _011.,206,CheckGroup(1); check line3 does not
have more than 1
exten = _011.,207,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}; use line-3
exten = _011.,307,Busy; Play busy if all lines
already used
so that the three lines will be choosen random, but still only one
user per line.
Can you give me a hint?
BTW, I have not tested the lines yet, ... if you spot an error, please
point it out.
bye
Ronald
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Re: [Asterisk-Users] Flashpannel: How to get more than 28 buttons?
You can share them here: http://asterconf.hopto.org/ Mike - Original Message - From: Robert Rozman [EMAIL PROTECTED] To: Nicolás Gudiño [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 17, 2005 12:10 AM Subject: Re: [Asterisk-Users] Flashpannel: How to get more than 28 buttons? Hi, I'd also like to see alternative op_style.cfg. Can we establish some place to share them ? I've also one with smaller buttons (but will have to count them :-) ... Regards, Rob. - Original Message - From: Nicolás Gudiño [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 16, 2005 1:26 PM Subject: Re: [Asterisk-Users] Flashpannel: How to get more than 28 buttons? Hi Ronald, I have setup flash pannel, ... looks nice, but so far I could not configure it to get more than 4x7 buttons. I tried to make the buttons smaller, but than just the entire picture is smaller. What did you change in op_style.cfg? You can have literally hundred of buttons per screen, or multiple 'context' to split your buttons into several screens. I wll send you an alternate op_style.cfg with smaller buttons offlist. Regards, -- Nicolás Gudiño Buenos Aires - Argentina ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.7.3 - Release Date: 15/03/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Best DB
But Budwieser tastes like water to most Australian beer drinkers. (Now I'm in trouble!) Mike - Original Message - From: Chris Albertson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 18, 2005 11:48 AM Subject: RE: [Asterisk-Users] OT: Best DB What is the best truck? A recent survey finds that there are far more Ford Rangr pickup trucks on the road then there are Frightliner 18 wheelers In another survey we find that Chevy outnumbers Porche. Closer to home in the computer world, more people use MS Windows than Solaris. I think Budwieser outsells every other beer. In most organizations followers outnumber the leaders The poor will always outnumber the rich. Still interrested in that database poll? What's the best DB. First you must define best. After you do that the answer is easy. --- David Brodbeck [EMAIL PROTECTED] wrote: -Original Message- From: Steven Critchfield [mailto:[EMAIL PROTECTED] Top Deployed Databases poll shows following databases in use: SQL Server with 78%, Oracle - 55%, MySQL - 33% and PostgreSQL - 8%. I see they created this with Mysql, 78 + 55 + 44 + 8 = 185% I'm sure if you add in the others we would get to something around 300% deployment. Presumably some sites had more than one type of database in use. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.7.3 - Release Date: 15/03/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Help Site - cut down on Mailing List questions
This is a re-post as it was pointed out that I replied to a different thread instead of creating a new post. Sorry for the additional traffic. Mike Dear All, I understand the excitement surrounding a service like Asterisk, and how easy it is to jump in and ask a heap of questions. I also know how frustrating it can be dealing with a 200+ post per day mailing list as one of the question answerers. When I discovered Asterisk, I had a lot of study to do, because there are no real-world examples out there, just the trivial ones on the tiki and in the manual. I hope to propose a solution. I have (in a small time) downloaded and set up a repositor where we should all post our conf files, in an effort to get a big resource of a lot of different setups that we know just work. The program is simple, and looks like crap and is a testiment to my programming skills (or lack thereof). If anyone feels like re-coding or hosting this, let me know. You can find this at: asterconf.hopto.org (i think this has popups for the free DNS) or home.exetel.com.au/azyc/asterconf In the same philosophy as the GPL and wiki, it is open to all to search, view and download the conf code, however to post and add new categories, you must register. The site will not send you any mail or spam or anything. Of course, you should scrub your conf files for IP addresses and user/secrets, but otherwise, please post as much as you like. Please also include a description of the purpose of the post, and what type of service it runs on, for better searching. As a registered user, you are also free to add comments to other people's code snippets (but not change the code), and add more categories and sub-categories. I have started by creating categories for the most common conf files, under both working and broken sections. As a new (or old) asterisk user, if you are stuck, feel free to post your conf in the broken section and hopefully someone will come to help you. This will stop people posting codes to this list and flodding it. If we all use this resource, the we will reduce the amount of posts for people looking for instant setups, who don't want to use AMP or otherwise. That way, we can return this list to the discussion of Asterisk issues, rather than just a startup resource and helpdesk. I'm always interested in anyone's comments. Cheers Mike Sander sanderm at iprimus.com.au +61 2 401 010 289 (Australian mobile) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dock-n-talk connection to asterisk
Hi Peter. Look in last weeks (1/3/05) Sydney Morning Herald Tuesday IT liftout. They talk there about GSM gateways. It was made by Ericson I think, for around $1000. It's not meant for computer, rather as a FXO/FXS gateway to plug your house phone in for exactly the purpose you are talking about. Of course, if it is a FXO gateway, I'm sure a RJ cable (possibly crossover) will plug it in to a TD400 Digium card nicely to get what you want. I'm interested to know your progress, I have a few clients also interested in Sydney. Cheers Mike - Original Message - From: Peter Illmayer [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, March 05, 2005 2:06 PM Subject: [Asterisk-Users] Dock-n-talk connection to asterisk Hi ALL I'm looking for feedback on how well this unit integrates into asterisk via an ata. Is the audio quality any good as thats the first thing to upset the wife if its no good. I'm looking for a reasonably priced GSM gateway 1800mhz for use in Australia that works with an ata. Quite happy to import something that works well... Currently PSTN to mobile is $0.40c per minute and going to a selected provider, it will only cost $0.05c per minute so the savings are enormous for me, hence my interest in the DOck-n-Talk Any feedback would be very much appreciated ! -- Open WebMail Project (http://openwebmail.org) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.6.2 - Release Date: 4/03/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Help Site - cut down on Mailing List questions
Dear All, I understand the excitement surrounding a service like Asterisk, and how easy it is to jump in and ask a heap of questions. I also know how frustrating it can be dealing with a 200+ post per day mailing list as one of the question answerers. When I discovered Asterisk, I had a lot of study to do, because there are no real-world examples out there, just the trivial ones on the tiki and in the manual. I hope to propose a solution. I have (in a small time) downloaded and set up a repositor where we should all post our conf files, in an effort to get a big resource of a lot of different setups that we know just work. The program is simple, and looks like crap and is a testiment to my programming skills (or lack thereof). If anyone feels like re-coding or hosting this, let me know. You can find this at: asterconf.hopto.org (i think this has popups for the free DNS) or home.exetel.com.au/azyc/asterconf In the same philosophy as the GPL and wiki, it is open to all to search, view and download the conf code, however to post and add new categories, you must register. The site will not send you any mail or spam or anything. Of course, you should scrub your conf files for IP addresses and user/secrets, but otherwise, please post as much as you like. Please also include a description of the purpose of the post, and what type of service it runs on, for better searching. As a registered user, you are also free to add comments to other people's code snippets (but not change the code), and add more categories and sub-categories. I have started by creating categories for the most common conf files, under both working and broken sections. As a new (or old) asterisk user, if you are stuck, feel free to post your conf in the broken section and hopefully someone will come to help you. This will stop people posting codes to this list and flodding it. If we all use this resource, the we will reduce the amount of posts for people looking for instant setups, who don't want to use AMP or otherwise. That way, we can return this list to the discussion of Asterisk issues, rather than just a startup resource and helpdesk. I'm always interested in anyone's comments. Cheers Mike Sander [EMAIL PROTECTED] +61 2 401 010 289 (Australian mobile) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Manager API - multi Originate cal ls
I'm sure this has been said, but the [EMAIL PROTECTED] installation of Flash Operator Panel shows the handset shaking when a phone is ringing, so there is a way to do it. I'd search in there. Mike - Original Message - From: mattf [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, March 03, 2005 6:29 AM Subject: RE: [Asterisk-Users] Asterisk Manager API - multi Originate cal ls Well, I'm not sure about the current release as I have not tested this, but on older releases for RBS T1s you would get a manager event showing a RING state. As for PRI, SIP and IAX2 I'm not sure, this is an inconsistent feature that differes depending on what kind of trunk you are using and what network the person you are calling is on. The only sure thing you can tell is a call pickup in all cases, ringing is much harder to detect. MATT--- -Original Message- From: Thomas Miller [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 02, 2005 2:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Manager API - multi Originate cal ls Hi Matt, in your experience is there a 100% reliable way to know that the callee phone is ringing? In my situation I don't need to know if they pick up or not, I need to know (as reliably as possible) if the calee phone number is ringing. Thanks, Tom --- mattf [EMAIL PROTECTED] wrote: ActionID does not return in all events related to an Action sent, sometimes it will just send you a success message and nothing more. Just try Originating a call from a meetme room over an outside line. You will get about 150 lines of output and only one message will have the ActionID in it, the success message. On the other hand the callerID is placed on many more of the events in the output. It is still the case that if you do complex Manager Actions, the ONLY solution for tracking a call is to use a custom CallerID. Action: Originate Exten: 8600080 Channel: local/[EMAIL PROTECTED] Context: default Priority: 1 Callerid: DF345678901234567890 Actionid: AID45678901234567890 MATT--- -Original Message- From: Bill Seddon [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 02, 2005 8:06 AM To: Stephen Owen hosted; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Manager API - multi Originate calls read in places that you use originate command and wait for an event back, does that mean you cannot place another originate until the event comes back ? Not in my experience. Originate will not send an event to the caller until either the intended caller (that is the extension used in Originate) has picked up their phone or a timeout occurs because the intended caller does not pick up their phone. You can send as many originate requests as you like but they will fail if more than one uses the same extension at the same time. The issue you will face is determining which event generated by Asterisk belongs to which origination request. For this reason, the Manager API allows you to specify an ActionID on any command. An ActionID is any string of characters that you use to uniquely identify each command use issue. Asterisk will include the ActionID with each related event so you know which events to respond to and which to ignore. You will see many events generated by Asterisk only some of which will relate to your command. The others will be events that Asterisk raises (for example when a phone registers) or events in response to commands issues by other Manager API users and at the command line. Take a look at Nicolas Gudino's Flash Operator Panel ( www.asternic.org http://www.asternic.org/ ) as it used the manager API extensively (albeit through a proxy) and will typically make many requests via the Manger API. Is it true that multiple API connections to Asterisk Manager API will crash it (thinking of alternative way to crack the nut) Again, not in my experience. Lyquidity Solutions Limited +44 (0) 208 241 0500 _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Owen hosted Sent: March 02, 2005 12:28 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk Manager API - multi Originate calls Been researching connecting over TCP\IP to Asterisk Manager API to initiate several concurrent calls to dial out. Prefer not to generate ASCII .call files. Question : I read in places that you use originate command and wait for an event back, does that mean you cannot place another originate until the event comes back ? Is it true that multiple API connections to Asterisk Manager API will crash it (thinking of alternative way to crack the nut) All help would be welcome - thanks Stephen Owen sip:[EMAIL PROTECTED] IM:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com
Re: [Asterisk-Users] Autodetecting faxes
That's all very well, but what do you do if you only have SIP extensions and IAX trunk - no Zaptel card. Will Fax detection still work at all? Thanks Mike - Original Message - From: Adrian Chapman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 08, 2005 8:24 PM Subject: Re: [Asterisk-Users] Autodetecting faxes Michael Welter wrote: Changing the order of things in extensions.conf around a smidge got it all working nicely :- [inbound-from-pstn] include = default exten = s,1,Answer exten = s,2,Wait,1 exten = s,3,Playback(thank-you-for-calling-please-wait-a-moment) exten = fax,1,Macro(faxreceive) exten = s,4,Do the normal phone call gubbins Is the position of the fax extension, between priorities 3 and 4, significant? What does 'show dialplan' display for the fax extension? It's there as much for flow readability as anything... The change of order was as much referring to moving the Playback forward from the voice handling macro, to give * time to hear the fax beep. Show Dialplan gives :- In each of my inbound call contexts 'fax' = 1. Macro(faxreceive) [pbx_config] No other mention at all. -- Adrian Chapman Director Trivas Ltd Business on the Move Mobility - Messaging - Infrastructure - Security - Remote Access 07796 690210 - 01582 626552 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.6 - Release Date: 7/02/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to download CVS with attended transfers
Hi I know that attended transfers are only available in the CVS Head. I downloaded the asterisk-update.sh script from voip-info.com and ran it with these parameters ./asterisk-update.sh update dev It looked as tho CVS HEAD was downloading and compiling, although it couldn't download the addons. However, now it's up and running, only blind transfers work with #, and I cannot change the blind transfer key to ##, it only takes the first character. And Attended transfers still isn't running. Is there something I've missed. The version info reports: Asterisk CVS-v1-0-02/03/05-10:24:22 Any help would be great. Thanks Mike -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.4 - Release Date: 1/02/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to download CVS with attended transfers
Hi I know that attended transfers are only available in the CVS Head. I downloaded the asterisk-update.sh script from voip-info.com and ran it with these parameters ./asterisk-update.sh update dev It looked as tho CVS HEAD was downloading and compiling, although it couldn't download the addons. However, now it's up and running, only blind transfers work with #, and I cannot change the blind transfer key to ##, it only takes the first character. And Attended transfers still isn't running. Is there something I've missed? The version info reports: Asterisk CVS-v1-0-02/03/05-10:24:22 Any help would be great. Thanks Mike -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.4 - Release Date: 1/02/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multi Asterisk Server Transfers
I believe this is what I have, but it still insists on running the transfer from the head office. Example: Provider --- IAX --- Head Office Provider --- SIP --- Remote Office Provider --- PSTN (Provider is the same * server in all cases) Call comes from PSTN to Head office. Head office transfers to 0 where is SIP extension according to Provider and 0 is to dial out on the trunk. Call is then connected as follows. PSTN - Provider - Head Office - Provider - Remote But after it is transferred, I want the resulting route to be: PSTN - Provider - Remote Otherwise Head office has 2 times the bandwidth running through it for a call not even going to one of it's own extensions. I had throught that the IAX connection between Provider and Head Office would pass off calls that way. Let me know, but thanks for all the help so far. Mike Instead I'd go for a co-located Asterisk that the remote SIP devices register with, and then link both * boxes (co-located and central office) using IAX2 with IAX native transfers enabled. Of course this means that the office * _only_ talks IAX and that all calls to the remote SIP clients _always_ go thru the co-located box (with its extra bandwidth). SER certainly is another way to go (as mentioned before), but in this specific setup I assume it complicates matters unnecessarily. Cheers, Philipp -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.1 - Release Date: 27/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multi Asterisk Server Transfers
Simple as that? Anyone know a good IAX phone (not softphone)? Thanks Mike Then you need to use the same protocol to the provider. One office is using SIP, the other is using IAX. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.1 - Release Date: 27/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multi Asterisk Server Transfers
Hi, We are in the business of setting up * servers for businesses, attached via IAX trunks to our VoIP provider (also using *). I have a client with a head office * server, who wants a number of remote offices, with just 1 SIP connection to each. I can arrange this no probs with our providers, but there are issues with transfer. I don't want the remote offices making their direct SIP connection to the head office, because bandwidth is limited and then for them to make an outgoing call, the head office has both an incoming and outgoing connection - or double the bandwidth. This is the same for an incoming call to head office that gets transferred to the remote, the call stays with the head office * server, and the server makes another outgoing call to the remote office. All these calls are free, but use double the bandwidth. The question: The remote offices can make direct SIP connections to our provider. If the head office * transfers a call, then the server releases the call entirely back to the providers * server and calls from there. I.E. call in to head office from PSTN through the provider. Call gets transferred to the remote office. Head office could then unplug/burn/blowup their asterisk server without disrupting the call between the remote office and the PSTN network. Is this possible? Companies with multiple * servers in many remote office, surely have this system, to conserve bandwidth? How is the transfer made? Mostly we are using X-PRO systems/Grandstream, with the [EMAIL PROTECTED] basic release. Thanks Mike -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.5 - Release Date: 26/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multi Asterisk Server Transfers
I agree with you. If every office had a * server, it would be fine. i.e. Office 1 rings office 2, then gets transferred to office 3, then connection is direct from office 1 to 3, and 2 releases all contact. However, what if office 3 is a 1 person office, with just a single SIP phone connected to the VoIP provider. Full IAX trunking can do hand-offs quite simply, I think, but when the destination is a single SIP connection, things get messy. Is this relevant to your answer, because I'm a little confused now? With thanks Mike Seems strange to be handling multiple * servers over SIP and ignoring IAX2. I'd be inclined to trunk between offices over IAX2. In fact, I'd use and IAX2 based ITSP and then be able to hand off calls in a .reinvite fashion without all the messy port handling. In addition you save on bandwidth by trunking multiple calls over one IAX2 connection. Less IP overhead, between offices and to the ITSP. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.5 - Release Date: 26/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SetGroup and CheckGroup problems
I have a rather long dial plan, but it includes support for call waiting.
However, the setgroup checkgroup commands don't seem to be working. Can
anyone help on this one?
Excerpts are below. First exten-vm is dialed and then dial-new.
As I understand, priority 1 increments the active channels for the caller
and then in dial-new priority 8 increments for Arg3, or the Callee
extension. Problem is, that priority 9 always goes on to 10 (i.e. group
never is on-the-phone.
Am I missing something?
When ext201 dials 202, CLI shows:
-- Executing Macro(SIP/201-8571, exten-vm|202|202) in new stack
-- Executing SetGroup(SIP/201-8571, 201) in new stack
-- Executing SetMusicOnHold(SIP/201-8571, default) in new stack
-- Executing SetVar(SIP/201-8571, FROMCONTEXT=exten-vm) in new stack
-- Executing GotoIf(SIP/201-8571, 0?9:5) in new stack
-- Goto (macro-exten-vm,s,5)
-- Executing Macro(SIP/201-8571, dial-new|15|tr|202|202) in new
stack
-- Executing DBget(SIP/201-8571, CallForwardIm=CF/202) in new stack
-- DBget: varname=CallForwardIm, family=CF, key=202
-- DBget: Value not found in database.
-- Executing Goto(SIP/201-8571, s|4) in new stack
-- Goto (macro-dial-new,s,4)
-- Executing DBget(SIP/201-8571, DNDStatus=DND/202) in new stack
-- DBget: varname=DNDStatus, family=DND, key=202
-- DBget: Value not found in database.
-- Executing Goto(SIP/201-8571, s|8) in new stack
-- Goto (macro-dial-new,s,8)
-- Executing SetGroup(SIP/201-8571, 202) in new stack
I'll be most grateful for any assistance.
Thanks
Mike
[macro-exten-vm]
exten = s,1,SetGroup(${CALLERIDNUM})
exten = s,2,SetMusicOnHold(default)
exten = s,3,Setvar(FROMCONTEXT=exten-vm) exten =
s,4,GotoIf($[${CALLERIDNUM} = ${ARG2}]?9:5) ;check self-voicemail exten =
s,5,Macro(dial-new,${RINGTIMER},${DIAL_OPTIONS},${ARG2},${ARG1})
[macro-dial-new]
;now check if destination is on a call
exten = s,8,SetGroup(${ARG3})
exten = s,9,CheckGroup(1)
;go to 110 then 25 if on the phone (CW handler), go to 10 if not on the
phone
exten = s,110,Goto(s,25)
;line is clear, begin dial sequence
exten = s,10,Setvar(ChanType=${E${ARG3}}) ;Get the channel type
exten = s,11,Dial(${ChanType}/${ARG3},${ARG1},${ARG2})
Mike Sander
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[Asterisk-Users] SetGroup, CheckGroup
I have a rather long dial plan, but it includes support for call waiting.
However, the setgroup checkgroup commands don't seem to be working. Can
anyone help on this one?
Excerpts are below. First exten-vm is dialed and then dial-new.
As I understand, priority 1 increments the active channels for the caller
and then in dial-new priority 8 increments for Arg3, or the Callee
extension. Problem is, that priority 9 always goes on to 10 (i.e. group
never is on-the-phone.
Am I missing something?
When ext201 dials 202, CLI shows:
-- Executing Macro(SIP/201-8571, exten-vm|202|202) in new stack
-- Executing SetGroup(SIP/201-8571, 201) in new stack
-- Executing SetMusicOnHold(SIP/201-8571, default) in new stack
-- Executing SetVar(SIP/201-8571, FROMCONTEXT=exten-vm) in new stack
-- Executing GotoIf(SIP/201-8571, 0?9:5) in new stack
-- Goto (macro-exten-vm,s,5)
-- Executing Macro(SIP/201-8571, dial-new|15|tr|202|202) in new
stack
-- Executing DBget(SIP/201-8571, CallForwardIm=CF/202) in new stack
-- DBget: varname=CallForwardIm, family=CF, key=202
-- DBget: Value not found in database.
-- Executing Goto(SIP/201-8571, s|4) in new stack
-- Goto (macro-dial-new,s,4)
-- Executing DBget(SIP/201-8571, DNDStatus=DND/202) in new stack
-- DBget: varname=DNDStatus, family=DND, key=202
-- DBget: Value not found in database.
-- Executing Goto(SIP/201-8571, s|8) in new stack
-- Goto (macro-dial-new,s,8)
-- Executing SetGroup(SIP/201-8571, 202) in new stack
I'll be most grateful for any assistance.
Thanks
Mike
[macro-exten-vm]
exten = s,1,SetGroup(${CALLERIDNUM})
exten = s,2,SetMusicOnHold(default)
exten = s,3,Setvar(FROMCONTEXT=exten-vm)
exten = s,4,GotoIf($[${CALLERIDNUM} = ${ARG2}]?9:5) ;check self-voicemail
exten = s,5,Macro(dial-new,${RINGTIMER},${DIAL_OPTIONS},${ARG2},${ARG1})
[macro-dial-new]
;now check if destination is on a call
exten = s,8,SetGroup(${ARG3})
exten = s,9,CheckGroup(1)
;go to 110 then 25 if on the phone (CW handler), go to 10 if not on the
phone
exten = s,110,Goto(s,25)
;line is clear, begin dial sequence
exten = s,10,Setvar(ChanType=${E${ARG3}}) ;Get the channel type
exten = s,11,Dial(${ChanType}/${ARG3},${ARG1},${ARG2})
Mike Sander
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RE: [Asterisk-Users] Call Pickup
I have a rather long dial plan, but it includes support for call waiting.
However, the setgroup checkgroup commands don't seem to be working. Can
anyone help on this one?
Excerpts are below. First exten-vm is dialed and then dial-new.
As I understand, priority 1 increments the active channels for the caller
and then in dial-new priority 8 increments for Arg3, or the Callee
extension. Problem is, that priority 9 always goes on to 10 (i.e. group
never is on-the-phone.
Am I missing something?
When ext201 dials 202, CLI shows:
-- Executing Macro(SIP/201-8571, exten-vm|202|202) in new stack
-- Executing SetGroup(SIP/201-8571, 201) in new stack
-- Executing SetMusicOnHold(SIP/201-8571, default) in new stack
-- Executing SetVar(SIP/201-8571, FROMCONTEXT=exten-vm) in new stack
-- Executing GotoIf(SIP/201-8571, 0?9:5) in new stack
-- Goto (macro-exten-vm,s,5)
-- Executing Macro(SIP/201-8571, dial-new|15|tr|202|202) in new
stack
-- Executing DBget(SIP/201-8571, CallForwardIm=CF/202) in new stack
-- DBget: varname=CallForwardIm, family=CF, key=202
-- DBget: Value not found in database.
-- Executing Goto(SIP/201-8571, s|4) in new stack
-- Goto (macro-dial-new,s,4)
-- Executing DBget(SIP/201-8571, DNDStatus=DND/202) in new stack
-- DBget: varname=DNDStatus, family=DND, key=202
-- DBget: Value not found in database.
-- Executing Goto(SIP/201-8571, s|8) in new stack
-- Goto (macro-dial-new,s,8)
-- Executing SetGroup(SIP/201-8571, 202) in new stack
I'll be most grateful for any assistance.
Thanks
Mike
[macro-exten-vm]
exten = s,1,SetGroup(${CALLERIDNUM})
exten = s,2,SetMusicOnHold(default)
exten = s,3,Setvar(FROMCONTEXT=exten-vm) exten =
s,4,GotoIf($[${CALLERIDNUM} = ${ARG2}]?9:5) ;check self-voicemail exten =
s,5,Macro(dial-new,${RINGTIMER},${DIAL_OPTIONS},${ARG2},${ARG1})
[macro-dial-new]
;now check if destination is on a call
exten = s,8,SetGroup(${ARG3})
exten = s,9,CheckGroup(1)
;go to 110 then 25 if on the phone (CW handler), go to 10 if not on the
phone
exten = s,110,Goto(s,25)
;line is clear, begin dial sequence
exten = s,10,Setvar(ChanType=${E${ARG3}}) ;Get the channel type
exten = s,11,Dial(${ChanType}/${ARG3},${ARG1},${ARG2})
Mike Sander
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RE: [Asterisk-Users] SetGroup and CheckGroup problems
Excuse my continued denseness, but I'm still not getting the groups concept.
I have 1 IAX trunk allowing multiple incoming and outgoing calls, and about
10 SIP channels. I don't have any ZAP cards or channels configured. Is the
SetGroup command type intended mainly for Zaptel interfaces??
I changed the CheckGroup(1) command to GetGroupCount(${ARG3}) where ARG3 is
the extension being dialed. Then I added SetVar(GPCNT=${GROUPCOUNT}) so I
could see the value of GROUPCOUNT in the CLI debug.
The answer is usually 1, whether the destination is on a call or not. When
they are conferencing with 2 external calls, it shows 2, but there doesn't
seem to by rhyme or reason. It makes sense to me to show 1 when they are on
the call, 2 when they have 2 going etc, but if they aren't on any calls, it
should show 0.
Am I missing something here, I'm sure it's really obvious.
With thanks
Mike
Mike Sander wrote:
I have a rather long dial plan, but it includes support for call waiting.
However, the setgroup checkgroup commands don't seem to be working. Can
anyone help on this one?
Long story short: you cannot put a channel into two groups, unless you
add categories to your group names. Calling SetGroup multiple times
without category designators just replaces the channel's group each time
you call it.
However, you do not need to use SetGroup/CheckGroup to check a group's
status; you can use GetGroupCount to directly check any group you want,
even one that the current channel is not in. If you are using CVS HEAD,
you can even use GetGroupMatchCount to get counts of multiple groups
with similar names. Also in CVS HEAD, you can set OUTBOUND_GROUP before
calling Dial(), and the channels it creates to actually call the targets
will be automatically placed into that group, as if you had used
SetGroup on them (which you cannot do normally, since you can't run any
dialplan code on those channels).
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Version: 7.0.300 / Virus Database: 265.7.3 - Release Date: 24/01/2005
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RE: [Asterisk-Users] call return?
For me this worked straight out of the box with [EMAIL PROTECTED] 0.3 Mike Sander Operations Manager Suite 4 / 38-48 Waterloo St Surry Hills N.S.W 2010 Phone:(02) 8307 8877 Fax:(02)93182254 Mobile:0401 010 289 Email: [EMAIL PROTECTED] Website: www.corporatebankinginternational.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Polk Sent: Sunday, 23 January 2005 4:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] call return? Hi: Can any one point me in the rite direction on this? I am using asterisk at home for learning purposes. I am trying to get the triditional *69 working. Has there been any success in getting it to announce the number and get it to give you the option to call back? Chris - Original Message - From: Diego Ventrice [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, January 22, 2005 8:03 AM Subject: Re: [Asterisk-Users] softswitch dilemma Thanks for answering Chad, Actually, I just want to Switch traffic between wholesale providers (my customers) which actually terminate traffic (or not, some of them have just controllers-softswitches like the one Im willing to set up) collect CDRs and bill them =) I have no gateways of my own (of any kind) so Im not originating nor terminating calls, just switching traffic is my goal, all this people use h.323 of course. Any advice would be appreciated. Thanks for your help D. Date: Fri, 21 Jan 2005 22:23:58 -0600 (CST) From: Chad Whitten [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] softswitch dilemma To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 are you looking to do actual pstn to voip termination? if so, then you are gonna need ss7, cama and imt trunks - things which asterisk doesnt necessarily support. now if you just want to buy pri/t1 from the local telco and sell voip services off an asterisk server that gets back to the pstn over these pri/t1's, then yes, asterisk can do this. Diego Ventrice said: Hello everybody, Im new to the list and also new to asterisk, Im wondering if I could set up asterisk as a softswitch, I guess for what I've been reading that It could be possible but almost all the info and documentation Ive found so far is about asterisk as a PBX, etc. Im willing to set a small voip wholesale traffic bussiness and Im not quite sure asterisk is the right chocie for that. An asterisk-ser or an asterisk-vocal combination may be the answer ? Thanks in advance for any help. Diego -- Chad Whitten ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UPS for Asterisk
I'd considering an UPS backup system for my Asterisk server. I understand this is a linux issue, not a * issue, except for the following... Is the harddisk activity on a standard asterisk install such that I don't really have to worry if the power cuts?? As I understand, if HD activity is minimal, the probability of HD failure is significantly reduced. P.S. Power regulation is not needed, only protection against instantaneous power loss. Mike Sander -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems transferring calls - HELP!
Hi. I have the following weird phenomenon on a standard * @ home installation. I use X-PRO on 2 or 3 computers. Blind Transfers: With incoming calls: I click transfer then dial extension the click transfer again. Result: Hangs up on incoming caller. With outgoing calls: I click transfer then dial extension the click transfer again. Result: Transfer is ok, dialed extension connects to outgoing call. Assisted Transfers: Place incoming caller on hold. Get new line and dial extension. Chat with extension then click transfer and the line number of incoming caller. Result: Incoming caller can hear new extension ok, but new extension can only hear music. OR Place incoming caller on hold. Get new line and dial extension. Chat with extension then place him on hold too. Return to incoming caller, click transfer and the line number of dialed extension. Result: Incoming caller can hear music only, but new extension can incoming caller. This is very perplexing. It is like the XPro is interacting with * in a way that it is transferring the MOH channel, not the person. And the order is weird. Transferring 1 to 2 gives reverse result to transferring 2 to 1. When I had individual SIP accounts to our VoIP provider's * server, rather than our own, everything worked fine. Please help if you can, this is baking my noodle!!! Mike Sander Operations Manager Suite 4 / 38-48 Waterloo St Surry Hills N.S.W 2010 Phone:(02) 8307 8877 Fax:(02)93182254 Mobile:0401 010 289 Email: [EMAIL PROTECTED] Website: www.corporatebankinginternational.com -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.1 - Release Date: 19/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems transferring calls - Part 2!
Ok. I've done more research and testing and here are the details. It is using the dialparties.agi (http://www.sprackett.com/asterisk/) file to dial . Originally with the dial options tr. I changed the options to Tt but no change. Transferring internal extensions between each other works fine. Example: 201 calls 202. 201 transfers 202 to 203. Transferring the IAX trunk to other internal is weird, as per my previous email. Example: DID calls 201. 201 transfers DID to 202. DID is either hungup or half connected (DID gets connected to 202, but 202 only hears music. DID can hear 202, even though 202 is hearing music). At the moment I can only transfer trunk calls through the parking system, which is a pain to teach people about... I'm really stumped on this one. Mike Sander Operations Manager Suite 4 / 38-48 Waterloo St Surry Hills N.S.W 2010 Phone:(02) 8307 8877 Fax:(02)93182254 Mobile:0401 010 289 Email: [EMAIL PROTECTED] Website: www.corporatebankinginternational.com -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.1 - Release Date: 19/01/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.1 - Release Date: 19/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DIDs anywhere but here?
We have DID's in 5 Australian cities for $5 per month. Mike Sander Operations Manager Suite 4 / 38-48 Waterloo St Surry Hills N.S.W 2010 Phone:(02) 8307 8877 Fax:(02)93182254 Mobile:0401 010 289 Email: [EMAIL PROTECTED] Website: www.corporatebankinginternational.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk Sent: Tuesday, 18 January 2005 3:46 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] DIDs anywhere but here? Are there affordable DIDs (preferably IAX) available anywhere outside the US? I want to use it to meet ICANN requirements for providing a valid phone number, yet pre-empting some of the telemarketing calls my domain registrations generate. (Yes, I asked a similar question about 900# availability before). I'd prefer to have a number somewhere outside the NANP, preferably an asian country. This number will (obviously) be low-volume (minutes/month at the most), and shouldn't cost more than a couple of bucks. Maybe a list member knows and/or is using one? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.0 - Release Date: 17/01/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.0 - Release Date: 17/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transferring calls on Asterisk with X-Lite
I am having trouble transferring calls using asterisk. I think it is my * installation, because this worked fine with the same system when it was hosted at our VoIP providers. I receive a call on my IAX Trunk, to my extensions. I speak to the incoming call and tell them Ill just transfer the call. I click Transfer, dial extension and click Transfer again. Normally the call will disappear on my system and start ringing on the new extension. In this case, the call just hangs up. Any Ideas? Do you have to setup transfers in the extensions at all? Does this have something to do with the Reinvite status of the SIP phones? With Thanks Mike Sander Operations Manager Suite 4 / 38-48 Waterloo St Surry HillsN.S.W 2010 Phone:(02) 8307 8877 Fax:(02)93182254 Mobile:0401 010 289 Email: [EMAIL PROTECTED] Website: www.corporatebankinginternational.com -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.0 - Release Date: 17/01/2005 image001.jpg___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] On Hold music
Do you have Zaptel cards installed? You need to have a timer installed (whatever that means). If you dont have a zaptel card, then use ztdummy to fake one. You need to download and compile the zaptel drivers (from asterisk website). Edit the makefile and find the line: TZOBJS=zonedata.lo tonezone.lo LIBTONEZONE=libtonezone.so.1.0 MODULES=zaptel tor2 torisa wcusb wcfxo wcfxs \ ztdynamic ztd-eth wct1xxp wct4xxp wcte11xp # ztdummy #MODULES+=wcfxsusb Then remove the # before ztdummy Type Make All Type Make Install Add a line to load the module ztdummy on boot using the /etc/rc.d files The command is modprobe ztdummy More information at: http://www.voip-info.org/tiki-print.php?page=Asterisk+timer+ztdummy Mike Sander Operations Manager Suite 4 / 38-48 Waterloo St Surry Hills N.S.W 2010 Phone:(02) 8307 8877 Fax:(02)93182254 Mobile:0401 010 289 Email: [EMAIL PROTECTED] Website: www.corporatebankinginternational.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Computer Onsite Support Sent: Tuesday, 18 January 2005 12:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] On Hold music Can anyone of you help me out with this issue. My Asterisk is working fine except my music-on-hold will NOT work even though I just retry three different other machines with different board and sound. [Manny Teixeira] al Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Manjit Riat Sent: Monday, January 17, 2005 8:42 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP URL for incoming I want to set up my asterisk to receive SIP calls using the URL [EMAIL PROTECTED] . I have my own DNS server but would like know what entry goes into it as I have never set up SRV records before. (if it matter it is a BIND dns server). thanx -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.0 - Release Date: 17/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk@Home systems
I am having trouble setting up Meetme with this CD. I have the latest which
was posted on sourceforge about 2-3 days ago. It seems to come with meetme
8200 and 8201 rooms, but I am getting invalid messages.
Can anyone help.
The Meetme.conf is:
conf = 8200
conf = 8201
The extensions are:
exten = _8XXX,1,Answer
exten = _8XXX,2,Wait(1)
exten = _8XXX,3,GotoIf($[${CALLERIDNUM} = ${EXTEN:1}]?5:4)
exten = _8XXX,4,MeetMe(${EXTEN}|sM)
exten = _8XXX,5,MeetMe(${EXTEN}|asM)
The extensions set up are 200 and 201.
I assume you dial 8200 to be administrator of your own meetme room.
Help if anyone knows please.
Thanks
Mike
--
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Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 265.6.11 - Release Date: 12/01/2005
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[Asterisk-Users] Limit outgoing trunk calls
Hi, We have an IAX provider that limits incoming IAX trunk calls, based on how many lines you purchase, but gives unlimited outgoing calls. I want to use the local Asterisk server to limit the outgoing number of calls, to retain high bandwidth. I.E. If we can only support 10 symultaneous high-quality calls on our broadband connection, I want the 11th person that dials the outgoing line extension to get a congestion/busy signal. Does Asterisk have a way of tracking how many people are on the trunk at one time, and accept/reject new calls based on that? Is it though the dialplan or the iax.conf? Thanks Mike Sander Operations Manager Suite 4 / 38-48 Waterloo St Surry HillsN.S.W 2010 Phone:(02) 8307 8877 Fax:(02)93182254 Mobile:0401 010 289 Email: [EMAIL PROTECTED] Website: www.corporatebankinginternational.com -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.6.11 - Release Date: 12/01/2005 image001.jpg___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
